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henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001/*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#ifndef MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
12#define MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000013
kwiberg2d0c3322016-02-14 09:28:33 -080014#include <memory>
henrik.lundin4cf61dd2015-12-09 06:20:58 -080015#include <string>
16
Danil Chapovalovb6021232018-06-19 13:26:36 +020017#include "absl/types/optional.h"
Fredrik Solenbergbbf21a32018-04-12 22:44:09 +020018#include "api/audio/audio_frame.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020019#include "modules/audio_coding/neteq/audio_multi_vector.h"
20#include "modules/audio_coding/neteq/defines.h"
Henrik Lundin3ef3bfc2018-04-10 15:10:26 +020021#include "modules/audio_coding/neteq/expand_uma_logger.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020022#include "modules/audio_coding/neteq/include/neteq.h"
23#include "modules/audio_coding/neteq/packet.h" // Declare PacketList.
24#include "modules/audio_coding/neteq/random_vector.h"
25#include "modules/audio_coding/neteq/rtcp.h"
26#include "modules/audio_coding/neteq/statistics_calculator.h"
27#include "modules/audio_coding/neteq/tick_timer.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020028#include "rtc_base/constructormagic.h"
29#include "rtc_base/criticalsection.h"
30#include "rtc_base/thread_annotations.h"
Mirko Bonadei71207422017-09-15 13:58:09 +020031#include "typedefs.h" // NOLINT(build/include)
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000032
33namespace webrtc {
34
35// Forward declarations.
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000036class Accelerate;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000037class BackgroundNoise;
38class BufferLevelFilter;
39class ComfortNoise;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000040class DecisionLogic;
41class DecoderDatabase;
42class DelayManager;
43class DelayPeakDetector;
44class DtmfBuffer;
45class DtmfToneGenerator;
46class Expand;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000047class Merge;
henrik.lundin91951862016-06-08 06:43:41 -070048class NackTracker;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000049class Normal;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000050class PacketBuffer;
ossua70695a2016-09-22 02:06:28 -070051class RedPayloadSplitter;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000052class PostDecodeVad;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000053class PreemptiveExpand;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000054class RandomVector;
55class SyncBuffer;
56class TimestampScaler;
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000057struct AccelerateFactory;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000058struct DtmfEvent;
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000059struct ExpandFactory;
60struct PreemptiveExpandFactory;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000061
62class NetEqImpl : public webrtc::NetEq {
63 public:
henrik.lundin55480f52016-03-08 02:37:57 -080064 enum class OutputType {
65 kNormalSpeech,
66 kPLC,
67 kCNG,
68 kPLCCNG,
69 kVadPassive
70 };
71
Henrik Lundinc417d9e2017-06-14 12:29:03 +020072 enum ErrorCodes {
73 kNoError = 0,
74 kOtherError,
75 kUnknownRtpPayloadType,
76 kDecoderNotFound,
77 kInvalidPointer,
78 kAccelerateError,
79 kPreemptiveExpandError,
80 kComfortNoiseErrorCode,
81 kDecoderErrorCode,
82 kOtherDecoderError,
83 kInvalidOperation,
84 kDtmfParsingError,
85 kDtmfInsertError,
86 kSampleUnderrun,
87 kDecodedTooMuch,
88 kRedundancySplitError,
89 kPacketBufferCorruption
90 };
91
henrik.lundin1d9061e2016-04-26 12:19:34 -070092 struct Dependencies {
93 // The constructor populates the Dependencies struct with the default
94 // implementations of the objects. They can all be replaced by the user
95 // before sending the struct to the NetEqImpl constructor. However, there
96 // are dependencies between some of the classes inside the struct, so
97 // swapping out one may make it necessary to re-create another one.
ossue3525782016-05-25 07:37:43 -070098 explicit Dependencies(
99 const NetEq::Config& config,
100 const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
henrik.lundin1d9061e2016-04-26 12:19:34 -0700101 ~Dependencies();
102
103 std::unique_ptr<TickTimer> tick_timer;
104 std::unique_ptr<BufferLevelFilter> buffer_level_filter;
105 std::unique_ptr<DecoderDatabase> decoder_database;
106 std::unique_ptr<DelayPeakDetector> delay_peak_detector;
107 std::unique_ptr<DelayManager> delay_manager;
108 std::unique_ptr<DtmfBuffer> dtmf_buffer;
109 std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator;
110 std::unique_ptr<PacketBuffer> packet_buffer;
ossua70695a2016-09-22 02:06:28 -0700111 std::unique_ptr<RedPayloadSplitter> red_payload_splitter;
henrik.lundin1d9061e2016-04-26 12:19:34 -0700112 std::unique_ptr<TimestampScaler> timestamp_scaler;
113 std::unique_ptr<AccelerateFactory> accelerate_factory;
114 std::unique_ptr<ExpandFactory> expand_factory;
115 std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory;
116 };
117
118 // Creates a new NetEqImpl object.
henrik.lundin@webrtc.orgea257842014-08-07 12:27:37 +0000119 NetEqImpl(const NetEq::Config& config,
henrik.lundin1d9061e2016-04-26 12:19:34 -0700120 Dependencies&& deps,
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000121 bool create_components = true);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000122
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200123 ~NetEqImpl() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000124
125 // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
126 // of the time when the packet was received, and should be measured with
127 // the same tick rate as the RTP timestamp of the current payload.
128 // Returns 0 on success, -1 on failure.
Henrik Lundin70c09bd2017-04-24 15:56:56 +0200129 int InsertPacket(const RTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800130 rtc::ArrayView<const uint8_t> payload,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000131 uint32_t receive_timestamp) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000132
henrik.lundinb8c55b12017-05-10 07:38:01 -0700133 void InsertEmptyPacket(const RTPHeader& rtp_header) override;
134
henrik.lundin7a926812016-05-12 13:51:28 -0700135 int GetAudio(AudioFrame* audio_frame, bool* muted) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000136
kwiberg1c07c702017-03-27 07:15:49 -0700137 void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
138
kwibergee1879c2015-10-29 06:20:28 -0700139 int RegisterPayloadType(NetEqDecoder codec,
henrik.lundin4cf61dd2015-12-09 06:20:58 -0800140 const std::string& codec_name,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000141 uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000142
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000143 int RegisterExternalDecoder(AudioDecoder* decoder,
kwibergee1879c2015-10-29 06:20:28 -0700144 NetEqDecoder codec,
henrik.lundin4cf61dd2015-12-09 06:20:58 -0800145 const std::string& codec_name,
kwiberg342f7402016-06-16 03:18:00 -0700146 uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000147
kwiberg5adaf732016-10-04 09:33:27 -0700148 bool RegisterPayloadType(int rtp_payload_type,
149 const SdpAudioFormat& audio_format) override;
150
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000151 // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
152 // -1 on failure.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000153 int RemovePayloadType(uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000154
kwiberg6b19b562016-09-20 04:02:25 -0700155 void RemoveAllPayloadTypes() override;
156
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000157 bool SetMinimumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000158
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000159 bool SetMaximumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000160
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000161 int LeastRequiredDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000162
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200163 int SetTargetDelay() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000164
Henrik Lundinabbff892017-11-29 09:14:04 +0100165 int TargetDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000166
henrik.lundin9c3efd02015-08-27 13:12:22 -0700167 int CurrentDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000168
henrik.lundinb3f1c5d2016-08-22 15:39:53 -0700169 int FilteredCurrentDelayMs() const override;
170
Henrik Lundin1ff41eb2018-06-21 12:36:28 +0000171 // Sets the playout mode to |mode|.
172 // Deprecated.
173 // TODO(henrik.lundin) Delete.
174 void SetPlayoutMode(NetEqPlayoutMode mode) override;
175
176 // Returns the current playout mode.
177 // Deprecated.
178 // TODO(henrik.lundin) Delete.
179 NetEqPlayoutMode PlayoutMode() const override;
180
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000181 // Writes the current network statistics to |stats|. The statistics are reset
182 // after the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000183 int NetworkStatistics(NetEqNetworkStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000184
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000185 // Writes the current RTCP statistics to |stats|. The statistics are reset
186 // and a new report period is started with the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000187 void GetRtcpStatistics(RtcpStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000188
Steve Anton2dbc69f2017-08-24 17:15:13 -0700189 NetEqLifetimeStatistics GetLifetimeStatistics() const override;
190
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000191 // Same as RtcpStatistics(), but does not reset anything.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000192 void GetRtcpStatisticsNoReset(RtcpStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000193
194 // Enables post-decode VAD. When enabled, GetAudio() will return
195 // kOutputVADPassive when the signal contains no speech.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000196 void EnableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000197
198 // Disables post-decode VAD.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000199 void DisableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000200
Danil Chapovalovb6021232018-06-19 13:26:36 +0200201 absl::optional<uint32_t> GetPlayoutTimestamp() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000202
henrik.lundind89814b2015-11-23 06:49:25 -0800203 int last_output_sample_rate_hz() const override;
204
Danil Chapovalovb6021232018-06-19 13:26:36 +0200205 absl::optional<CodecInst> GetDecoder(int payload_type) const override;
kwiberg6f0f6162016-09-20 03:07:46 -0700206
Danil Chapovalovb6021232018-06-19 13:26:36 +0200207 absl::optional<SdpAudioFormat> GetDecoderFormat(
ossuf1b08da2016-09-23 02:19:43 -0700208 int payload_type) const override;
kwibergc4ccd4d2016-09-21 10:55:15 -0700209
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200210 int SetTargetNumberOfChannels() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000211
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200212 int SetTargetSampleRate() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000213
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000214 // Flushes both the packet buffer and the sync buffer.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000215 void FlushBuffers() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000216
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000217 void PacketBufferStatistics(int* current_num_packets,
218 int* max_num_packets) const override;
turaj@webrtc.org7df97062013-08-02 18:07:13 +0000219
henrik.lundin48ed9302015-10-29 05:36:24 -0700220 void EnableNack(size_t max_nack_list_size) override;
221
222 void DisableNack() override;
223
224 std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
turaj@webrtc.orgff43c852013-09-25 00:07:27 +0000225
henrik.lundin114c1b32017-04-26 07:47:32 -0700226 std::vector<uint32_t> LastDecodedTimestamps() const override;
227
228 int SyncBufferSizeMs() const override;
229
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000230 // This accessor method is only intended for testing purposes.
henrike@webrtc.org47658f12014-09-10 22:14:59 +0000231 const SyncBuffer* sync_buffer_for_test() const;
minyue5bd33972016-05-02 04:46:11 -0700232 Operations last_operation_for_test() const;
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000233
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000234 protected:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000235 static const int kOutputSizeMs = 10;
minyue5bd33972016-05-02 04:46:11 -0700236 static const size_t kMaxFrameSize = 5760; // 120 ms @ 48 kHz.
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000237 // TODO(hlundin): Provide a better value for kSyncBufferSize.
minyue17461792016-05-03 13:32:05 -0700238 // Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for
239 // calculating correlations of current frame against history.
240 static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000241
242 // Inserts a new packet into NetEq. This is used by the InsertPacket method
243 // above. Returns 0 on success, otherwise an error code.
244 // TODO(hlundin): Merge this with InsertPacket above?
Henrik Lundin70c09bd2017-04-24 15:56:56 +0200245 int InsertPacketInternal(const RTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800246 rtc::ArrayView<const uint8_t> payload,
ossu17e3fa12016-09-08 04:52:55 -0700247 uint32_t receive_timestamp)
danilchap56359be2017-09-07 07:53:45 -0700248 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000249
henrik.lundin6d8e0112016-03-04 10:34:21 -0800250 // Delivers 10 ms of audio data. The data is written to |audio_frame|.
henrik.lundin@webrtc.orge1d468c2013-01-30 07:37:20 +0000251 // Returns 0 on success, otherwise an error code.
henrik.lundin7a926812016-05-12 13:51:28 -0700252 int GetAudioInternal(AudioFrame* audio_frame, bool* muted)
danilchap56359be2017-09-07 07:53:45 -0700253 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000254
255 // Provides a decision to the GetAudioInternal method. The decision what to
256 // do is written to |operation|. Packets to decode are written to
257 // |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
258 // DTMF should be played, |play_dtmf| is set to true by the method.
259 // Returns 0 on success, otherwise an error code.
260 int GetDecision(Operations* operation,
261 PacketList* packet_list,
262 DtmfEvent* dtmf_event,
danilchap56359be2017-09-07 07:53:45 -0700263 bool* play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000264
265 // Decodes the speech packets in |packet_list|, and writes the results to
266 // |decoded_buffer|, which is allocated to hold |decoded_buffer_length|
267 // elements. The length of the decoded data is written to |decoded_length|.
268 // The speech type -- speech or (codec-internal) comfort noise -- is written
269 // to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
270 // comfort noise, those are not decoded.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000271 int Decode(PacketList* packet_list,
272 Operations* operation,
273 int* decoded_length,
274 AudioDecoder::SpeechType* speech_type)
danilchap56359be2017-09-07 07:53:45 -0700275 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000276
minyuel6d92bf52015-09-23 15:20:39 +0200277 // Sub-method to Decode(). Performs codec internal CNG.
danilchap56359be2017-09-07 07:53:45 -0700278 int DecodeCng(AudioDecoder* decoder,
279 int* decoded_length,
minyuel6d92bf52015-09-23 15:20:39 +0200280 AudioDecoder::SpeechType* speech_type)
danilchap56359be2017-09-07 07:53:45 -0700281 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
minyuel6d92bf52015-09-23 15:20:39 +0200282
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000283 // Sub-method to Decode(). Performs the actual decoding.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000284 int DecodeLoop(PacketList* packet_list,
minyuel6d92bf52015-09-23 15:20:39 +0200285 const Operations& operation,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000286 AudioDecoder* decoder,
287 int* decoded_length,
288 AudioDecoder::SpeechType* speech_type)
danilchap56359be2017-09-07 07:53:45 -0700289 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000290
291 // Sub-method which calls the Normal class to perform the normal operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000292 void DoNormal(const int16_t* decoded_buffer,
293 size_t decoded_length,
294 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700295 bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000296
297 // Sub-method which calls the Merge class to perform the merge operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000298 void DoMerge(int16_t* decoded_buffer,
299 size_t decoded_length,
300 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700301 bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000302
303 // Sub-method which calls the Expand class to perform the expand operation.
danilchap56359be2017-09-07 07:53:45 -0700304 int DoExpand(bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000305
306 // Sub-method which calls the Accelerate class to perform the accelerate
307 // operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000308 int DoAccelerate(int16_t* decoded_buffer,
309 size_t decoded_length,
310 AudioDecoder::SpeechType speech_type,
Henrik Lundincf808d22015-05-27 14:33:29 +0200311 bool play_dtmf,
danilchap56359be2017-09-07 07:53:45 -0700312 bool fast_accelerate)
313 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000314
315 // Sub-method which calls the PreemptiveExpand class to perform the
316 // preemtive expand operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000317 int DoPreemptiveExpand(int16_t* decoded_buffer,
318 size_t decoded_length,
319 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700320 bool play_dtmf)
321 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000322
323 // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
324 // noise. |packet_list| can either contain one SID frame to update the
325 // noise parameters, or no payload at all, in which case the previously
326 // received parameters are used.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000327 int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
danilchap56359be2017-09-07 07:53:45 -0700328 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000329
330 // Calls the audio decoder to generate codec-internal comfort noise when
331 // no packet was received.
minyuel6d92bf52015-09-23 15:20:39 +0200332 void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
danilchap56359be2017-09-07 07:53:45 -0700333 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000334
335 // Calls the DtmfToneGenerator class to generate DTMF tones.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000336 int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
danilchap56359be2017-09-07 07:53:45 -0700337 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000338
Henrik Lundin1ff41eb2018-06-21 12:36:28 +0000339 // Produces packet-loss concealment using alternative methods. If the codec
340 // has an internal PLC, it is called to generate samples. Otherwise, the
341 // method performs zero-stuffing.
342 void DoAlternativePlc(bool increase_timestamp)
343 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
344
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000345 // Overdub DTMF on top of |output|.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000346 int DtmfOverdub(const DtmfEvent& dtmf_event,
347 size_t num_channels,
danilchap56359be2017-09-07 07:53:45 -0700348 int16_t* output) const
349 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000350
351 // Extracts packets from |packet_buffer_| to produce at least
352 // |required_samples| samples. The packets are inserted into |packet_list|.
353 // Returns the number of samples that the packets in the list will produce, or
354 // -1 in case of an error.
Peter Kastingdce40cf2015-08-24 14:52:23 -0700355 int ExtractPackets(size_t required_samples, PacketList* packet_list)
danilchap56359be2017-09-07 07:53:45 -0700356 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000357
358 // Resets various variables and objects to new values based on the sample rate
359 // |fs_hz| and |channels| number audio channels.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000360 void SetSampleRateAndChannels(int fs_hz, size_t channels)
danilchap56359be2017-09-07 07:53:45 -0700361 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000362
363 // Returns the output type for the audio produced by the latest call to
364 // GetAudio().
danilchap56359be2017-09-07 07:53:45 -0700365 OutputType LastOutputType() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000366
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000367 // Updates Expand and Merge.
368 virtual void UpdatePlcComponents(int fs_hz, size_t channels)
danilchap56359be2017-09-07 07:53:45 -0700369 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000370
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000371 // Creates DecisionLogic object with the mode given by |playout_mode_|.
danilchap56359be2017-09-07 07:53:45 -0700372 virtual void CreateDecisionLogic() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000373
pbos5ad935c2016-01-25 03:52:44 -0800374 rtc::CriticalSection crit_sect_;
danilchap56359be2017-09-07 07:53:45 -0700375 const std::unique_ptr<TickTimer> tick_timer_ RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800376 const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
danilchap56359be2017-09-07 07:53:45 -0700377 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800378 const std::unique_ptr<DecoderDatabase> decoder_database_
danilchap56359be2017-09-07 07:53:45 -0700379 RTC_GUARDED_BY(crit_sect_);
380 const std::unique_ptr<DelayManager> delay_manager_ RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800381 const std::unique_ptr<DelayPeakDetector> delay_peak_detector_
danilchap56359be2017-09-07 07:53:45 -0700382 RTC_GUARDED_BY(crit_sect_);
383 const std::unique_ptr<DtmfBuffer> dtmf_buffer_ RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800384 const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
danilchap56359be2017-09-07 07:53:45 -0700385 RTC_GUARDED_BY(crit_sect_);
386 const std::unique_ptr<PacketBuffer> packet_buffer_ RTC_GUARDED_BY(crit_sect_);
ossua70695a2016-09-22 02:06:28 -0700387 const std::unique_ptr<RedPayloadSplitter> red_payload_splitter_
danilchap56359be2017-09-07 07:53:45 -0700388 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800389 const std::unique_ptr<TimestampScaler> timestamp_scaler_
danilchap56359be2017-09-07 07:53:45 -0700390 RTC_GUARDED_BY(crit_sect_);
391 const std::unique_ptr<PostDecodeVad> vad_ RTC_GUARDED_BY(crit_sect_);
392 const std::unique_ptr<ExpandFactory> expand_factory_
393 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800394 const std::unique_ptr<AccelerateFactory> accelerate_factory_
danilchap56359be2017-09-07 07:53:45 -0700395 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800396 const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
danilchap56359be2017-09-07 07:53:45 -0700397 RTC_GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000398
danilchap56359be2017-09-07 07:53:45 -0700399 std::unique_ptr<BackgroundNoise> background_noise_ RTC_GUARDED_BY(crit_sect_);
400 std::unique_ptr<DecisionLogic> decision_logic_ RTC_GUARDED_BY(crit_sect_);
401 std::unique_ptr<AudioMultiVector> algorithm_buffer_
402 RTC_GUARDED_BY(crit_sect_);
403 std::unique_ptr<SyncBuffer> sync_buffer_ RTC_GUARDED_BY(crit_sect_);
404 std::unique_ptr<Expand> expand_ RTC_GUARDED_BY(crit_sect_);
405 std::unique_ptr<Normal> normal_ RTC_GUARDED_BY(crit_sect_);
406 std::unique_ptr<Merge> merge_ RTC_GUARDED_BY(crit_sect_);
407 std::unique_ptr<Accelerate> accelerate_ RTC_GUARDED_BY(crit_sect_);
408 std::unique_ptr<PreemptiveExpand> preemptive_expand_
409 RTC_GUARDED_BY(crit_sect_);
410 RandomVector random_vector_ RTC_GUARDED_BY(crit_sect_);
411 std::unique_ptr<ComfortNoise> comfort_noise_ RTC_GUARDED_BY(crit_sect_);
412 Rtcp rtcp_ RTC_GUARDED_BY(crit_sect_);
413 StatisticsCalculator stats_ RTC_GUARDED_BY(crit_sect_);
414 int fs_hz_ RTC_GUARDED_BY(crit_sect_);
415 int fs_mult_ RTC_GUARDED_BY(crit_sect_);
416 int last_output_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
417 size_t output_size_samples_ RTC_GUARDED_BY(crit_sect_);
418 size_t decoder_frame_length_ RTC_GUARDED_BY(crit_sect_);
419 Modes last_mode_ RTC_GUARDED_BY(crit_sect_);
420 Operations last_operation_ RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700421 size_t decoded_buffer_length_ RTC_GUARDED_BY(crit_sect_);
422 std::unique_ptr<int16_t[]> decoded_buffer_ RTC_GUARDED_BY(crit_sect_);
423 uint32_t playout_timestamp_ RTC_GUARDED_BY(crit_sect_);
424 bool new_codec_ RTC_GUARDED_BY(crit_sect_);
425 uint32_t timestamp_ RTC_GUARDED_BY(crit_sect_);
426 bool reset_decoder_ RTC_GUARDED_BY(crit_sect_);
Danil Chapovalovb6021232018-06-19 13:26:36 +0200427 absl::optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(crit_sect_);
428 absl::optional<uint8_t> current_cng_rtp_payload_type_
danilchap56359be2017-09-07 07:53:45 -0700429 RTC_GUARDED_BY(crit_sect_);
430 uint32_t ssrc_ RTC_GUARDED_BY(crit_sect_);
431 bool first_packet_ RTC_GUARDED_BY(crit_sect_);
Henrik Lundin1ff41eb2018-06-21 12:36:28 +0000432 NetEqPlayoutMode playout_mode_ RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700433 bool enable_fast_accelerate_ RTC_GUARDED_BY(crit_sect_);
434 std::unique_ptr<NackTracker> nack_ RTC_GUARDED_BY(crit_sect_);
435 bool nack_enabled_ RTC_GUARDED_BY(crit_sect_);
436 const bool enable_muted_state_ RTC_GUARDED_BY(crit_sect_);
437 AudioFrame::VADActivity last_vad_activity_ RTC_GUARDED_BY(crit_sect_) =
henrik.lundin500c04b2016-03-08 02:36:04 -0800438 AudioFrame::kVadPassive;
henrik.lundinb1fb72b2016-05-03 08:18:47 -0700439 std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
danilchap56359be2017-09-07 07:53:45 -0700440 RTC_GUARDED_BY(crit_sect_);
441 std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_);
Henrik Lundin3ef3bfc2018-04-10 15:10:26 +0200442 ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
443 ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
minyue@webrtc.orgd7301772013-08-29 00:58:14 +0000444
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000445 private:
henrikg3c089d72015-09-16 05:37:44 -0700446 RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000447};
448
449} // namespace webrtc
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200450#endif // MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_